Voice communications are increasingly shifting to web and Internet based applications that are outside of traditional telephony networks. Enterprise users desire to access their unified communications applications with their own Internet connected mobile devices, and consumers increasingly prefer Internet based communications channels for accessing contact centers.
Some communications service providers (“CSPs”) and enterprises have deployed real-time communications (“RTC”) applications based on a protocol known as WebRTC. WebRTC is an open Internet standard for embedding real-time multimedia communications capabilities (e.g., voice calling, video chat, peer to peer (“P2P”) file sharing, etc.) into a web browser. For any device with a supported web browser, WebRTC can use application programming interfaces (“APIs”) to equip the device with RTC capabilities without requiring users to download plug-ins. By using WebRTC, CSPs may create new web based communications services and extend existing services to web based clients.
WebRTC communications are authenticated according to typical web authentication technologies and corresponding databases. However, some communications networks use authentication technologies and databases that are different than web authentication technologies and databases. This may cause authentication issues when a WebRTC applications needs to communicate with an entity in such communications networks.